From d343a25173e846fe2474700ed1849df4d74d88f3 Mon Sep 17 00:00:00 2001 From: Alexander Traud Date: Mon, 26 Oct 2015 17:42:03 +0100 Subject: [PATCH] chan_sip: Do not send all codecs on INVITE. Since version 13, Asterisk sent all allowed codecs as callee, even when the caller did not request/support them. In case of dynamic RTP payloads, this led to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the intersection between the requested and the supported codecs is send again. ASTERISK-24543 #close Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287 --- channels/chan_sip.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 8f76e9cc31..f0d4de53d6 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -13332,7 +13332,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int } /* Finally our remaining audio/video codecs */ - for (x = 0; x < ast_format_cap_count(p->caps); x++) { + for (x = 0; ast_test_flag(&p->flags[0], SIP_OUTGOING) && x < ast_format_cap_count(p->caps); x++) { tmp_fmt = ast_format_cap_get_format(p->caps, x); if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {